commit | d8a1b7a5c5e2f71f34a0d635c2593cf6c55de26d | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Thu Dec 06 12:00:27 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Dec 06 12:37:27 2018 |
tree | a70c7c0ba2d3a657eb840029d8c9fccbbd4bda87 | |
parent | 5658ea660e470affa83fb23c3f40c4bb1e0b79d9 [diff] |
Use opaque int as payload_type in MediaTransportInterface Replaces enum VideoCodecType for video frames and uint8_t for audio frames. Also delete method MediaTransportVideoSinkInterface::OnKeyFrameRequested; it needs to be added as a send-side interface instead (for a later cl). Bug: webrtc:9719 Change-Id: I2cfdbacc267afc75c448512e2cc6de0ec9966a2d Reviewed-on: https://webrtc-review.googlesource.com/c/113180 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25918}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.